similar to: CID passthrough

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the

Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth

We would like implement G729 passthrough for our calls and get rid of the encoding overhead, and a little confused as to how to do this, and some unanswered questions. Do we need the open source G729? If so, do we still need the patent license. Not so much of an issue, just checking. Finally, a recent howto of how to enforce pci passthrough and disable encoding would be greatly appreciated. Oh,

Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried

I have several providers that do NOT allow using your own CID. On one of these, I suppress outgoing CID simply because I don't want people calling us back on that number. When I dial out through this SIP provider to an incoming number of another SIP provider (for testing dialplan stuff, since there's no good reason to call myself and go through 2 systems) I see an incoming CID of

Hi all, using Skype for Asterisk I have the following problem. In my dialplan I need to have a CID matching (example.skype) over a DID (test.skype) : [example] exten => test.skype/example.skype,1, NoOp(nothing) exten => test.skype/example.skype,n, Hangup() Where test.skype and example.skype are Skype business account. In this case, when I get a : CLI> show dialplan example I get: [

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Hi All, Sorry for the double post, but I'm in a real bind. I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version CVS-v1-0-11/09/04-12:27:27. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and

Quick and dirty too hungover to think/search... ;) Sorry list AsteriskBoxA -> SBC/PSTN voodoo -> world wide Interweb -> AsteriskBoxB Account on BoxA UsernameJohn<12125551212> Account on BoxB UsernameJohn<102> They're both the same users, had to do some funky trunking (managed firewall provider is playing not in my backyard games)... So anyway, I need to take specific

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

Hi Guys, I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and populate into the * CDR fine. I connected to a DMS-500, NI2 (national) protocol and

The latest libpri is supposed to handle this properly, but doesn't seem to. Here's the debug info. CALLERID(name) is set to empty. < Protocol Discriminator: Q.931 (8) len=66 < TEI=0 Call Ref: len= 2 (reference 256/0x100) (Sent from originator) < Message Type: SETUP (5) < [04 03 80 90 a2] < Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability:

Dear I am using a2billing accounting software, how can I charge on the destination target not at the caller side Ex: if user A have 10$ and user B have 10$ ,and the onnet call charge cost 1$ When user A call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$ Regards ********************************************* No employee or agent is authorized to conclude

I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What

Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

lol just posted this yesterday, it's for any ODBC DSN so Access or SQL or an Excel spreadsheet, as long as it's set as a DSN. This will work with outgoing Caller ID as well, it's just how you set it up in your dialplan. If you want I can email you the .agi since email will undoubtedly mangle the script. hth. HOWTO: A simple AGI application to modify incoming CallerID on the fly using

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

Hi The first step I have to go through when users call into our IVR is to handle the case where users' PBX hides their CID number. In that case, I need to have them type their phone number (ten digits). OTOH, those who call without hiding their CID number are sent directly to the main menu. How would I go about prompting users for their phone number? Here's what I have at this point:

I'm thinking of doing an app to work with the CID that's gotten from the Zap channel. All the CID's I see from within the US are 10 digit numbers. I'm out in the rural areas of the US, and no-one ever calls me from overseas. If they did, what would the CID look like? What does the CallerID look like overseas? How many countries provide it? murf